DunedinDragon

I'm assuming you'll feed that monitor out of the 1/4" mono out of the pod? I've used fx-150 in the past, but that was for vocals. 150watts may seem a lot, but they can be overwhelmed pretty quickly if your stage volume gets too high. My biggest question is will the rest of the band be able to hear you? On a stand my guess is it will work pretty well, but I'd strongly advise you to take your POD to the music store and play through it to get a feel for how it will handle your output. Since it's geared for vocals I'm not sure how full range the EQ is, but you'll get a sense of that when you play through it. It might require a couple of adjustments in your Global EQ to get it to sound right. But let us know how it works out.

I personally don't worry about it. The only time it would make a difference from my understanding is when the recording has a very wide dynamic range such as in the recording of an orchestra. In typical recording situations of most modern music you're not likely to see any benefit, particularly given the additional storage and processing it requires. As long as it sounds good to my ears and to people that play it, I'm good with it. Especially given the garbage speakers and headphones most people use nowdays to listen to music. It's just another one of those distractions that takes your focus away from doing the only thing that makes any real difference, actually getting better at playing and performing music.

This is becoming clear you have a defective unit. I also have a POD HD Pro X desktop unit. I've inadvertantly left it on all night after I went to bed and have never had any heat issue at all with the unit. In fact the unit NEVER gets warm. Given the recent cutting out of the sound it seems like there's no choice but to get it back to Line 6 and have it looked at. I'm assuming it's still under warranty?

In all likelihood you've either deleted or turned off the pedal in the signal chain. Try a different preset and see if it works elsewhere. But first, sit down and read the Pilot's Guide. It won't take that long and you need to know a few things about the navigation system before anyone can lead you through any of this stuff. It could be any number of things but unless we can be sure you understand the directions we give you I'm not sure we can be of much help.

rd2rk - Theoretically any analog signal is subject to interference and potential loss of signal over long distances, so it's certainly a possibility. Low impedence XLR connections are less subject to such things, but the potential is always there. That's the reason for SPDIF. If it were that significant a possibility however, SPDIF would be a much more prominent connection than it is in actual usage and you'd likely see PA loudspeakers for live performances using them. But in reality and actual usage in the industry it's not considered very significant so that's why you don't see it all that often. You do see it used more often in recording environments however due to the potential for electromagnetic interference caused by equipment. Even there you don't see it used that often in my experience. Granted in a home environment it could be useful in that a fair amount of homes have electromagnetic anomoles due to dirty or unconditioned power. That's one of the reasons I'm a big advocate of using power conditioners for equipment used for recording in home studios. If it were to be a real problem you'd likely hear more than just a loss of signal. More likely you'd hear slight buzzing, humming or some other noise artifact. That's what typically causes someone to opt for SPDIF instead of analog connections. As I mentioned in my post sound quality isn't a consideration in my case for why I opt to use a separate audio interface. It has more to do with configurability of the preamp stage for recording as well as the ability to not have to have the POD connected into the computer in order to have sound through my monitors or headsets. Very often when I'm mixing or mastering I don't have any need for the POD and I'd rather it be free for other uses. It's also comes in handy when I'm working out arrangements while I play back tracks from the DAW and work out guitar parts to them on the POD. I let the audio interface manage the sound coming from the DAW and from the POD leaving the POD free to be dedicated to processing the patches with no risk of latency.

You're probably experiencing the same as what I experience on my Yamaha DXR12 in that, as a PA loudspeaker, it's designed to cover the full range of frequences pretty evenly as compared to the guitar cab setting which will replicate the limited response of a guitar cabinet. In order to cut out the non natural guitar frequencies I use the Global EQ and put the Low Cut at about 85 HZ and I put the High Cut frequency at around 7400 HZ. You'd have to experiment on your L3t for the right frequencies for what you want to hear, but I would imagine somewhere in those areas you'll see that you'll get rid of the muddy and boomy bass response as well as the piercing frequencies at the very high end and retain the clarity and articulation you want. However, I just noticed you said you're using the RCA inputs. If you're using those for backing tracks you'll have to be careful about adjusting the Global EQ so it doesn't diminish the fidelity of the backing track which is likely to use most if not all the range of the speakers in the PA setting.

I think people can convince their ears of anything that want to hear, whether it's logical or not. Mathmatically speaking, once you convert analog to it's digital representation, the uncoversion won't won't restore it to the pure analog representation. Rather it will be the analog representation of the digital source. Any additional conversion to digital of that analog representation will simply reproduce the same 1's and 0's of the original digital source representation since digital operations are commutative in nature as long as the sampling rate remains the same.

My patches are individually geared toward the guitar I use on them. Not only because the tone is different, but the volume can be considerably different depending on the pickups. That way I only have to gain stage any single patch when we have sound checks at a gig and the rest will fall right in line. The only adjustment I ever make to the Master Volume is during sound check and gain staging.

Even though the POD will work perfectly fine as an audio interface I choose to have a separate audio interface (MOTO UltraLite MK3) that I plug my POD into via XLR the same as I would any recording studio board or mixer. Typically stand alone audio interfaces have a few more features that allow for better control over both recording and playback, more appropriate connections for studio monitors, not to mention better support from the manufacturer since that's their main business. More importantly I don't have to worry if I take the POD somewhere for a gig or practice and it's out of the house for a few days. The sound system on the computer still works the same as always.

If you use unbalancd TS 1/4" cables into a balanced 1/4" TRS input, it should work fine, but the signal will be unbalanced and will be subject to noise and interference. If you use TRS balanced cables in an unbalanced output (like the POD) it will also be unbalanced as the ring part of the cable will be ignored. You can buy a direct box that converts your unbalanced 1/4" signal into at balanced 1/4" TRS signal.

It just seems odd there are so many problems people seem to experience with the audio interface from the POD. I've used a variety of audio interfaces over the years from a variety of manufacturers, some more stable than others depending on how well the drivers are written. I suppose much of it could be a lack of memory or processing power on the platforms given most of the users don't specify. It just gives me pause about the stability of the drivers provided by L6 since these can vary greatly from vendor to vendor. While I would agree that running other foreground tasks and applications could be an issue when recording, buffer size and number of buffers aside, general background tasks and services shouldn't typically interfere with normal ASIO driven processing on a normally functioning computer since it's a much lighter load than general WASAPI and all sound processing drivers run by election from the controlling application which places them in user space at a higher priority than system level processes.

A separate question here: is the "Line 6 Audio-MIDI device Driver" a different driver than the one installed as part of the POD Edit program? I personally don't use the POD as an audio interface as I have a MOTU UltraLite as my primary audio interface, but I'm curious about why there seem to be so many problems people keep encountering with it and the precautions you guys have been advising regarding the additional USB activities should I decide to try it some day.

I'd suggest you take this one problem at a time. If you're getting a blue screen on your PC you need to resolve that problem first regardless of the sound issues. The sound issues may be an artifact of whatever is causing the blue screen, or it may be a separate issue. Regardless either you have a problem with the driver, or you have some virus, or some other significant issue that's causing the blue screen. I'd start with a system restore and/or virus check on the PC and try to get it to the point that plugging or unplugging the POD doesn't cause a blue screen. You can't address other issues until the operating system is functioning correctly first.

True but I was talking more about the comparison between stereo speakers and loudspeakers. I've not come across any headphones targetted at that type of sonic response. Even top end headphones tend to be targetted more toward studio mixing which is quite different.

From what I can tell getting real FRFR headphones is a bit of a challenge. Even studio quality headphones ($300+) tend to try and simulate a audiophile stereo speaker setup rather than a pure FRFR loudspeaker, so there's going to be some coloration in the output. I suspect they could be useful in practicing, but I'm not sure I would trust them for setting up patches by themselves. You'd probably need to tweek the patches a bit after you connected to your normal output speakers.

2.0. I doubt that config is even available in 3.0

You need what most people refer to as a USB printer cable. Technically it's a regular (not mini or micro) USB A Male to a regular (not mini or micro) USB B Male connector.

I do quite a few Santana sounds and I tend to prefer the Fender Deluxe with the 112 speaker and 57 on axis as the core amp due to it's inherent warmth. I use a bit of gain on the amp but use a tube drive to add a bit of gain as well to the basic sound which is slightly crunchy/punchy/clean. Then I kick in a Tube Screamer for the leads with about 65% gain to get those nice singing sustain tones.

One thing I would encourage you to do in approaching the POD HD is to begin with learning some of the differences in the various amp models rather than just the EFX. Bear in mind the POD HD can be used as an EFX board, but it's real benefit comes from the combination of amp modeling along with the effects. I notice the setup you typically use could be used as a basis for creating your first patch to just get you started with learning where everything is and how to set them up as a patch and save it. Those are the basic functions you need to know to really begin exploring the capabilities. There are a;so a number of really good videos from Line6Support either on this site or on YouTube that will walk you through creating various patches such as a Brian May patch from Queen or a Jimmy Page Led Zep patch. Those videos are awesome for helping you through creating the various patches. I would suspect you would want to set the POD HD output to Studio/Direct take the 1/4" guitar mono output from the POD HD into your Powerengine60, and run the XLR output to the mixing board for the PA in a manner similar to what you do now. This is the way I set my rig up, but I actually use a full range powered speaker cabinet for my monitor. Once you master creating various patches using different amps you can begin to create custom patches that best suit specific styles of songs. For example, I use the Roland Jazz Rivet model from the vintage pack as the basis for 10,000 Reasons which gives me a nice ringing, clear tone; but for some things that are more upbeat with a little crunch to them I'll use something like a Fender Deluxe for a nice crunchy but warm sound, or the Marshall as a base amp for heavier rock sounds. That's when you'll discover the real power of this unit in creating a range of presets to match different songs rather than just one amp setting that changes based on effects. That's when this really becomes a lot of fun!!

I do think a lot of folks figure these units will be something of a "turn it on and it will do the magic" without understanding the learning curve. I've been doing this sort of thing for at least 6 years now and I still come across interesting new ways of creating a variety of tones or adjustments that are extraordinarily useful. I don't particularly care for downloading custom tones as I have done that in the past and haven't been very satisfied with the results. For the most part they might sound perfect on the guitar and rig that someone was using when they created it, but to adapt it to my specific guitar and rig takes almost as much time as it would to create one from scratch. What I do find very useful and informative is watching some of the instructional videos of how someone created a given patch. I'd much rather learn how they achieved it so I can apply some of the same approaches in patches I create for myself.

That wouldn't surprise me at all. The profile and response characteristics of a guitar amp is completely different from that of a FRFR speaker arrangement. I would imagine the difference you're hearing is from the restriction of frequency response in various areas of the frequency range, and the added coloration a guitar amp adds in other frequency areas. This is one of those either/or situations...you gotta choose one or the other. This is why I only use FRFR setups for setup, practice, and gigging.

I want to give a tip of the hat to Hughanico on his advice to use a Studio EQ at the end of the chain. I used that little trick today as I was setting up for a performance this weekend and that is by FAR the easiest way I've ever found to normalize volumes between the patches. Not to mention it's quite handy at compensating for the characteristics of different guitars that may need some adjustment in low or mid/high frequency gains on output. That's an awsome little tool for those final finishing touches on a patch to get it right where it needs to be both volume-wise and tone wise in a very clean way before sending it off to the FOH.

Actually you should buy for "need" rather than "want". Any feature can be made to sound important. Whether or not it really IS important to you is based on whether it addresses something that's been a limitation specific to you. That's what differentiates a "need" from a "want". For example I might be want a beautiful 3 bedroom house on a lake. But since it's just me and my dog and neither of us are interested in fishing, boating, or other water sports I don't "need" it because I likely woundn't take advantage of all it has to offer.

There's always going to be a next generation of something, and at some point it becomes useful to upgrade to it if it provides something you can't do with your current equipment. I will say I'm intrigued with the whole dual DSP the Helix has to offer because I can see where it might add a bit more realism to the modeling. However, I have to ask myself if I'm currently plagued by the need for a more realistic modeling environment, and I can't say that I am at this point. There may come a time when the state of modeling advances to the point that I need to consider it to keep up, but for right now I don't see the need. I never run into DSP limitations on my patches and I'm perfectly comfortable with the interface on the 500X, so given those are the key feature improvements I don't see it has anything to offer me right now.

Having done this with various modeling amps over the years, this is always a constant battle because amp models vary so much in the way they respond to volume and drive. The only consistent way I have been able to do it is by ear patch by patch. In time you get a feel for how each amp model is going to respond to volume and gain, so you can pretty much begin to predict where it will come out. But as was mentioned before, perceived volume and measured volume are very different, so I've been a lot more successful using my ears as a gauge. A patch that's overdriven will often be perceived as louder than a clean patch with more mellow effects such as a chorus even when they're pushing the same SPL. As I lay out my patches for a performance I switch between them from patch to patch (song to song) making minor adjustments in either the volume or drive on the amp, or within the effects like the compressors or overdrive pedals until they all to line up pretty well volume wise to my ears. I then save that patches so that when I need something like it again I already know it's pretty well gain staged. The end goal is to ideally set my master volume on the POD at a set level that blends with the rest of the band and I can gain stage with the FOH sound check on pretty much any of one patch and the rest of the patches will all come out pretty close.